Archives : January-2015
In some situations I got the following message on asterisk console:* Autodestruct on dialog 857128833@192.168.2.129 with owner SIP/1015-00000002 in place (Method:BYE). Rescheduling destruction for 10000 ms*I would like to raise a manager event, to t..
allI have a strange issue with 1.8.11.0 on a production Asterisk machine at our head office, and the same issue with a production machine at a branch office.Every now and then, on the head office machine, ODBC CEL and CDR logging will stop working…
All,The issue appearing at the random for confbridge module i.e. in some cases if a participant joins the confbridge, he/she unable to hear others which make him/herto hangup the call and redial the bridge again. By joining the bridge second time, particip..
Tomorrow night a few community services will have intermittent availability due to maintenance. This maintenance will begin at approximately 9:00 PM CST[1] and should last no longer than two hours, ending around 11:00 pM CST.The affected services ar..
allWe have recently upgraded Asterisk from 1.8.26 to 13.0.0 and now we have a whole lot of errors like this in our logs:[2015-01-05 06:43:36] ERROR[14636] stasis_cache.c: Attempting to remove an item from the SIP/ics0002-cached cache that isnt the..
I continued the developing of Openfire and Asterisk integration projects, and now Im here to invite you to test AstDemo, that allows VoIP operations directly in XMPP clients.So if use Openfire and wanna test AstDemo, please send me some feedback, suggesti..
I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact.I am having trouble mapping them to the typical SIP configuration settings on a phone.Suppose I have a phone with two line butto..
all,Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222@mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is
fellow asterisk users,Im trying to use feature application defined in application map.its defined as follows:lbxvml => 1,self/caller,Macro,Jump2VoicemailIts working properly when called party answers the call, but Id like to have feature usable wh..
happy new year!I am still trying to make T.38 work. In the meantime, I have upgraded to Asterisk 13.1.0, and I am using the most recent PJSIP library (compiling that stuff myself). My local fax software is capable of T.38, as is my ITSP; Asterisk s..